System and method for adaptive multi-rate (AMR) vocoder rate adaptation

ABSTRACT

The present invention includes a time-division-multiple-access (TDMA) communication system having a base station and at least one mobile station, each transmitting and receiving an analog radio-frequency signal carrying digitally coded speech. The speech is encoded using a vocoder which samples a voice signal at variable encoding rates. During periods when the radio-frequency channel is experiencing high levels of channel interference, the encoded voice channel having a lower encoding rate is chosen. This low-rate encoded voice is combined with the high degree of channel coding necessary to ensure reliable transmission. When the radio-frequency channel is experiencing low levels of channel interference, less channel coding is necessary and the vocoder having a higher encoding rate is used. The high-rate encoded voice is combined with the lower degree of channel coding necessary to ensure reliable transmission. The appropriate levels of channel coding necessary for reliable transmission are determined by various channel metrics, such as frame erase rate and bit error rate. The determination of the appropriate vocoder rate and level of channel coding for both the uplink and downlink may be determined centrally at the base station, with the vocoder rate and level of channel coding for the uplink being relayed to the mobile station. Alternatively, the appropriate vocoder rate and level of channel coding for the downlink may be determined by the mobile station, and the appropriate vocoder rate and level of channel coding for the uplink may be determined by the base station.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. Ser. No. 10/124,464, filed onApr. 16, 2002, now U.S. Pat. No. 7,164,710, which is a is a continuationof U.S. Ser. No. 09/080,013, filed on May 15, 1998, now U.S. Pat. No.6,529,730.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates generally to wireless communicationsystems. More particularly, the present invention relates to a wirelesscommunication system having an adaptive multi-rate (AMR) vocoder tomaximize the voice quality while minimizing the level of channel coding.

2. Description of the Related Art

As the use of wireless communication systems become increasinglypopular, a variety of methods are being developed to increase the numberof mobile communication devices a system can simultaneously service. TheGlobal System for Mobile Communications (GSM), also referred to as theGroup Speciale Mobile, is one example of a wireless communication systemwhich is constantly being adapted to increase the number of simultaneoususers.

The GSM system is modeled after standards created by the EuropeanTelecommunications Standards Institute (ETSI) and operates between atelecommunication base station (BS) and a mobile station (MS) using apair of frequency bands in a frequency division duplex (FDD)configuration. The first frequency band occupies the frequency spectrumbetween 890 to 915 Megahertz (MHZ), and the second frequency bandoccupies the frequency spectrum between 935 to 960 MHZ. Typically, thefirst frequency range is used for the lower power transmissions from theMS to the BS, and the second frequency range is used for the higherpower transmission from the BS to the MS. Each frequency range isdivided into 125 channels with 200 Kilohertz (kHz) spaced carrierfrequencies.

The GSM communication system is a time-division-multiple-access (TDMA)system. In the GSM TDMA system, each carrier frequency is divided intoeight (8) time slots. Because each MS is assigned a single time slot onone channel in both the first frequency range and the second frequencyrange, a total of 992 MS may use the BS at the same time.

A typical speech channel for GSM communication is sampled at 8 KHz andquantized to a resolution of 13 bits, providing for the digitization ofspeech ranging from 0-4 KHz by a voice encoder, also referred to as avocoder. The 13 bits are then compressed by a factor of eight (8) in afull-rate vocoder to a voice data digital bit stream of 13 kilobits persecond (Kbit/s). Because GSM uses a complex encryption technique withinterleaving and convolution coding, a high degree of system integrityand bit error control are achieved. In fact, despite multi-path andco-channel interference, the GSM system may continue to operate despitea carrier-to-interference ratio (C/I) as low as 9 dB, in comparison to atypical advanced mobile phone service (AMPS) analog system requiring amaximum C/I of 17 dB.

Depending upon the topography of an area, a typical BS may providecommunication services to any number of MSs within a radius up to 35Kilometers. Consequently, with the rising popularity of mobilecommunication devices, it is often the case that during peak periods ofuse, such as rush-hour traffic, all channels are fully occupied and theBS is not able to provide communication services all of the MS in itsregion.

In order to avoid the inability to service all MS within a region, theETSI has contemplated a modification of the GSM standard to increase thedensity of the communication channels. However, because the allocatedfrequency spectrum of 25 MHZ with 125 separate 200 KHz carrier channelsis fixed, a current approach to increasing the density of thecommunication system is to increase the number of users per channel. Ingeneral, this density increase is achieved by decreasing the amount ofdigital information which is sent to and from each BS, thereby allowingeach BS to support more users in a 200 kHz frequency band.

One approach to decreasing the amount of digital information passingbetween a BS and a MS is to decrease the vocoder rate of the digitalvoice data from a full-rate vocoder rate of 13 kilobits per second(Kbits/s) to a half-rate vocoder rate of 5.6 Kbits/s. Although theability currently exists to effectively double the number of users onany one communication channel from eight (8) to sixteen (16) by usingthe half-rate vocoder, it has been found that the 5.6 Kbits/s vocoderrate is barely acceptable as the speech quality is significantlydecreased.

In light of the above, it would be advantageous to provide acommunication system that provides for the user density of a half-ratevocoder system, while providing the voice quality approaching orexceeding that of a full-rate vocoder system. It would also beadvantageous to provide a communication system that provides for themodification of the communication channel to incorporate only the amountof channel coding necessary to achieve a reliable communication linkbetween the MS and the BS.

SUMMARY OF THE INVENTION

Broadly, the present invention provides for a wireless communicationsystem having the ability to increase or decrease the vocoder rate andchannel coding in response to the level of interference present on thewireless communication channel, resulting in a communication channelhaving the best possible speech quality. This may be accomplished ineither a full-rate or half-rate GSM communication system by decreasingthe amount of channel coding during periods of low channel interferenceto allow transmission of more speech information, representing a highervocoder rate and resulting in a higher speech quality. During periods ofhigher channel interference, the amount of channel coding may beincreased to the maximum channel coding allowed in a GSM communicationnetwork. This increased channel coding provides for consistent andreliable call handling, and results in a lower vocoder rate having alower speech quality.

In an embodiment of the present invention, atime-division-multiple-access (TDMA) communication system includes abase station (BS) and at least one mobile station (MS), eachtransmitting and receiving an analog radio-frequency signal carryingdigitally coded speech. The speech is digitally encoded using a vocoderwhich samples a voice signal at different encoding rates. Alternatively,the speech may be encoded using a number of different vocoderssimultaneously, with each vocoder having a different encoding rate.During periods when the radio-frequency channel is experiencing highlevels of channel noise or interference, the encoded voice channelhaving a lower encoding rate is chosen. This lower-rate encoded voice iscombined with the high degree of channel coding necessary to ensurereliable transmission. When the radio-frequency channel is experiencinglow levels of channel interference, less channel coding is necessary andthe vocoder having a higher encoding rate is used. The high-rate encodedvoice is combined with the lower degree of channel coding necessary toensure reliable transmission. The appropriate level of channel codingnecessary for reliable transmission is determined by various channelmetrics, such as frame erase rate and bit error rate.

The determination of the appropriate vocoder rate and level of channelcoding for both the uplink and downlink may be determined centrally atthe base station, with the vocoder rate and level of channel coding forthe uplink being relayed to the mobile station. Alternatively, theappropriate vocoder rate and level of channel coding for the downlinkmay be determined by the mobile station, and the appropriate vocoderrate and level of channel coding for the uplink may be determined by thebase station.

BRIEF DESCRIPTION OF THE DRAWINGS

The nature, objects, and advantages of the invention will become moreapparent to those skilled in the art after considering the followingdetailed description in connection with the accompanying drawings, inwhich like reference numerals designate like parts throughout, wherein:

FIG. 1 is a diagram of a typical wireless telecommunication system,including a base station and a number of mobile stations;

FIG. 2 is a schematic diagram of the hardware of a typical wirelesstransceiver of the present invention and includes three separatevocoders, each having a different vocoder rate;

FIG. 3 is a graph of the relative performance characteristics of awireless communication system implementing a variable vocoder rate;

FIG. 4 illustrates the coding, combination and interleaving of speechblocks into a frame, and the variation of the ratio of channel coding tospeech coding for various levels of radio-frequency channel noise andinterference;

FIG. 5 is a state diagram illustrating the change of vocoder rate basedupon the current status of the communication system, including the FEand BER metrics;

FIG. 6 depicts a sequence of steps which are performed in thecommunication system wherein the mobile station calculates the downlinkvocoder rate based on its calculations of a number of channel qualitymetrics;

FIG. 7 depicts a sequence of steps which are performed in thecommunication system wherein the mobile station forwards its channelquality metrics to the base station where the vocoder rate for thedownlink is determined and communicated to the mobile station;

FIG. 8 is a quantization table identifying the bits transmitted from themobile station to the base station in order to provide the base stationwith the necessary channel metric information to determine the mobilestations vocoder rate;

FIG. 9 is a quantization table identifying the received bits whichcorrespond to the channel quality metrics made by the mobile station;and

FIG. 10 depicts a sequence of steps which are performed in thecommunication system wherein the base station calculates the uplinkvocoder rate based on its calculations of a number of channel qualitymetrics.

DETAILED DESCRIPTION

System Architecture of a Preferred Embodiment

Referring first to FIG. 11 an exemplary communication system of thepresent invention is shown and generally designated 100. Communicationsystem 100 operates in compliance with the GSM communication standardwhich includes a time-division-multiple-access (TDMA) communicationscheme. In general, a TDMA communication system provides for thetransmission of two or more data channels over the same radio-frequencychannel by allocating separate time intervals for the transmission ofeach data channel. In a GSM system, each 200 kilohertz (kHz)radio-frequency channel is divided into repeating time frames, eachframe having a duration of 4.615 milliseconds. Each frame contains eight(8) time intervals (also called “slots”) each having a duration of 577microseconds (4,615/8) and assigned to a different user.

Communication system 100 includes a base station (BS) 102 which receivessignals from a mobile switching center (MSC) 106 via communicationchannel 108. This communication channel includes telephone and/ordigital information which may typically originate from land-basedtelephone systems. Base station 102 transmits information to, andreceives information from, mobile stations (MS) 110, 112, and 114 whichare within cell 120. Cell 120 is a geographical region within which allmobile stations communicate with the base station 102. Typically, thesecells range may have radii ranging from twenty-five (25) to thirty-five(35) kilometers, and may include such geographical disturbances such asbuildings 130 or mountains 132. As used herein, the term “information”shall be defined to include digital data, encrypted digital data,convolutionally coded, soft-coded, and/or hard-coded data, digital bitsor a bit stream, or any other data type known in the art.

Because a GSM-based communication system operates with paired frequencybands in a frequency-division-duplex (FDD) mode, base station (BS) 102sends information to the mobile station (MS) 110 over a firstradio-frequency channel 116, typically in the 890 to 915 MHZ range andreferred to as the “downlink,” and mobile station 110 sends informationto the base station 102 over a second radio-frequency channel 118,typically in the 935 to 960 MHZ range and referred to as the “uplink.”Although a GSM-based communication system operates using two frequencybands, it is nonetheless possible to implement the present invention ina system where both the BS and MS transmit and receive over the sameradio-frequency channel.

Communication system 100 may support a number of mobile stations (MSs)110, 112, and 114. In fact, under the GSM standard, each 25 MHZfrequency band is divided into 125 channels with 200 Kilohertz (KHz)spaced carrier frequencies. With each carrier frequency supporting eight(8) separate users, a single GSM communication system may support nearlyone thousand (1,000) simultaneous users.

Given the high possible number of simultaneous users contributing toco-channel interference, and the presence of atmospheric andgeographical sources of interference, there are periods of time duringwhich a considerable amount of channel noise and interference is presenton the communication system 100. Moreover, the presence of buildings 130and mountains 132 result in multi-path distortion which further degradethe reliability of transmissions through the communication system 100.Additionally, because each MS may be moving in a different directionwith respect to the BS, either towards or away from the BS at speeds upto 250 kilometers-per-hour (156 miles-per-hour), the possibility that acommunication link will be temporarily or permanently disrupted is evenhigher.

In an attempt to minimize the deleterious effects of channel noise andinterference on both the uplink and downlink communication channels, asignificant amount of channel coding is added to the digital voice data.Channel coding is generally defined to include the process of combiningthe encoded digital voice data from the vocoder, with any redundantdata, parity data, cyclic-redundant-checking (CRC) or other check datanecessary to ensure the reliable transmission of the voice data. Thecode rate is the ratio of data bits to total bits (k/n), and istypically just over one-half (½) in an ordinary GSM-based system withfull-rate vocoders, and just below one-half (½) in a system withhalf-rate vocoders.

During the channel coding process, the introduction of the necessaryerror correction, redundant data, parity data, CRC or check data isaccomplished by convolutionally coding the digital voice data from thevocoder, with the necessary channel coding data. This results in aconvolutionally encoded digital data stream which includes a mixture ofvoice data and channel coding. As will be more thoroughly discussed inconjunction with FIG. 2, this digital data stream is modulated andamplified for transmission over a radio-frequency channel. Uponreception of the modulated digital data stream, the data stream ischannel de-modulated and the voice data and channel coding isconvolutionally decoded and separated.

During periods with high levels of channel noise, the introduction ofsignificant channel coding provides for an increased reliability of thecommunication channel. On the other hand, a data stream containing asignificant amount of channel coding information limits the amount ofvoice data which can be transmitted, and during periods of low channelnoise, results in an inefficient use of the communication channel.Consequently, the present invention monitors the current level ofchannel noise, and either increases the amount of channel coding toimprove channel reliability, or decreases the amount of channel codingto provide for the transmission of more voice data.

Although the current GSM-based communication systems dictate a maximumvocoder rate of 13 Kbits/s for a full-rate vocoder system, the presentinvention contemplates a much higher maximum vocoder rate up to thebandwidth limitation of the wireless communication channel itself. Forinstance, if the current level of channel noise or interference isminimal, it is possible to provide a communication link having virtuallyno channel coding and thus providing for a voice data rate of 22Kbits/s. This would correspond to a voice bandwidth over 4 kHz,resulting in a voice channel having a frequency range and correspondingvoice quality well beyond that of a traditional 4 kHz voice bandwidth.

Transceiver Architecture

Referring now to FIG. 2, a circuit diagram of a transceiver of oneembodiment of the present invention is shown and generally designated200. The transmitter portion of circuit 200 includes a microphoneelement 202, such as an electret-type microphone, that receives anacoustic signal, such as a user's voice, and converts the acoustic voicesignal to an analog electrical signal. This analog electrical signalpasses through amplifier 204 for amplification and filtering, and issupplied to the inputs of three (3) separate voice encoders, or vocoders206, 208, and 210.

A vocoder is an analog-to-digital converter (ADC) which is createdespecially for the digital encoding and compression of analog voicedata. Vocoders are designed around high speed digital signal processors(DSPs) and use a form of linear predictive coding which is intended tomodel the human vocal chords in order to produce realistic syntheticspeech with the minimum of memory. In a GSM communication system withfull-rate vocoders, voice data is sampled at the rate of 8 kHz andquantized to a resolution of thirteen (13) bits and compressed to a bitrate of 13 Kbits/s. In a GSM communication system with half-ratevocoders, voice data is sampled at the rate of 8 kHz and quantized to aresolution of thirteen (13) bits and compressed to give a bit rate of5.6 Kbits/s.

In the present invention, vocoders 206, 208, and 210 each receives theamplified voice signal from amplifier 204 and each vocoder iscontinuously encoding the acoustic voice signals at different rates. Forexample, vocoder 206 may encode the voice signal at a vocoder rate of 8Kbits/s, vocoder 208 may encode the voice signal at a lower vocoder rateof 6 Kbits/s, and vocoder 210 may encode the voice signal at an evenlower vocoder rate of 4 Kbits/s. The particular vocoder rates discussedherein are merely exemplary, and it is to be appreciated that a vocoderof virtually any rate may be used, so long as the representative digitaldata rate is capable of being transmitted over the radio-frequencycommunication channel.

The outputs from vocoders 206, 208 and 210 are fed into switch 212 whichis controlled by processor 214 having a memory storage 215. Processor214 in the present embodiment is a microprocessor. However, processor214 may instead be any conventional single or multi-chippedmicroprocessor, digital signal processor, microcontroller, or any othersuitable digital processing apparatus known in the art. Memory storage215 in the present embodiment may include an electrically erasableprogrammable read-only-memory (EEPROM), read-only-memory (ROM),random-access-memory (RAM), diskettes or other magnetic recording media,optical storage media, or any combination thereof. Electronicinstructions for controlling the operation of processor 214, in the formof program code, may be stored in memory storage 215.

Based upon a predefined selection process, described in greater detailbelow, processor 214 determines the proper vocoder rate and selects theoutput of the appropriate vocoder 206, 208 or 210 for passage throughswitch 212 to encoder 216. For example, if the voice signal is to beencoded at a full-rate of 8 Kbits/s, then the output of vocoder 206would be selected by processor 214 and passed through switch 212.Alternatively, if the voice is to be encoded at the rate of 6 Kbits/s,the output of vocoder 208 would be selected. Encoder 216 receives thedigital voice data from the vocoder and adds the level of channel codingcorresponding to the vocoder rate selected.

Once passed through encoder 216, the now-encoded digital voice data ismixed with the analog output of voltage-controlled-oscillator (VCO) 218to modulate the digital voice data onto a carrier frequency in modulator220. Modulator 220 modulates a gaussian-minimum-shift-key (GMSK) signalon a radio-frequency carrier which is then passed through variable poweramplifier 222 and through transmit/receive switch 224 to antenna 226 fortransmission. A GMSK signal incorporates gaussian-shaped pulses and isintended improve the resilience of the communication channel toco-channel interference. As an alternative to GMSK, other modulationmethods known in the art may be used, such as BPSK, QPSK, or FSK.

Control of transmit/receive switch 224 is accomplished by processor 214in a method well known in the art. In single antenna transceivers, it isoften necessary to switch the antenna between the transmitting andreceiving portions of the circuitry in order to isolate the sensitivereceiver electronics from the higher power signal generated by thetransmitter.

The receiver portion of circuit 200 begins with antenna 226 whichreceives an analog radio-frequency signal that is passed throughtransmit/receive switch 224 to intermediate-frequency (IF) amplifier andmixer 240. Mixer 240 removes the carrier frequency from theradio-frequency signal and passes the remaining analog signal to ananalog-to-digital converter (ADC) 242. ADC 242 converts the receivedanalog signal to a digital signal which is then passed throughequalization block 244 where the digital signal may be filtered and thedigital bit stream recovered, and to channel decoder 246.

As will be discussed in more detail below, processor 214 receives asignal, in the form of rate bits, from channel decoder 246. These ratebits identify the appropriate vocoder rate needed to decode the currentvoice data encoded in the received signal. Based upon the rate bits,processor 214 selects vocoder 250, 252, or 254 using switch 248, and thedigital voice data from the decoded radio-frequency channel is passedfrom channel decoder 246, through switch 248, and to the appropriatevocoder 250, 252, or 254. For example, if the digital voice data wasencoded at a full-rate of 8 Kbits/s, then processor 214 would operateswitch 248 to send the digital voice data to vocoder 250 which, in thepresent embodiment, decodes at the full-rate of 8 Kbits/s. Vocoder 250decodes the digital information received from channel decoder 246, andre-creates the original analog voice signal which is then passed throughamplifier 256 and out speaker 258 to be heard by the user.

In an alternative embodiment, vocoders 206, 208 and 210 of circuit 200may be replaced by a single vocoder (shown by dashed lines 270) havingmultiple encoding rates, or a variable encoding rate. For instance, thesingle variable-rate vocoder 270 may be capable of encoding acousticsignals from amplifier 204 at rates between 4 Kbits/s, and 8 Kbits/s asdetermined by processor 214. Similarly, vocoders 250, 252, and 254 maybe replaced by a single variable vocoder 272.

Although the present invention is discussed in conjunction with a TDMAcommunication system, it is to be appreciated that the use of a TDMAcommunication scheme is merely exemplary, and the present invention maybe practiced on any number of alternative communications systems, suchas code-division-multiple-access (CDMA) andfrequency-division-multiple-access (FDMA), for example.

Referring now to FIG. 3, a graphical representation of systemperformance is shown and generally designated 300. Graph 300 includes avertical axis 302 labeled “Voice Quality” and a horizontal axis 304labeled “Carrier-to-Noise Ratio (C/N).” As discussed herein, the termC/N is considered to include a carrier-to-interference (C/I) portion. Insummary, graph 300 represents the performance of communication systemsbased on the level of channel coding and corresponding vocoder rates.More specifically, three separate curves are shown and each representsthe performance of a particular system configuration. For example, graph306 represents the performance of a communication system using afull-rate vocoder, with a minimum level of channel coding. As can beenseen, curve 306 begins at a higher initial voice quality, but as the C/Ndecreases, the level of interference due to the lower level of channelcoding eventually causes a marked decrease in the voice quality.

Similarly, curve 312 represents the performance of a communicationsystem using a mid-rate vocoder with a corresponding mid-level ofchannel coding. Such a mid-rate vocoder rate could be 6 Kbits/s.Although the initial voice quality shown by curve 312 is maintained fora longer period, it too suffers from the interference caused by thelower level of channel coding.

Finally, curve 318 represents the system performance of a communicationsystem using a low-rate vocoder with a corresponding higher level ofchannel coding. In this case, the high level of channel coding providesfor a continuous communication link despite a significant decrease inthe C/N, however, the voice quality is lower than either the systemshown by curve 306 or 312.

In order to maintain the highest level of voice quality possible,despite the decreasing C/N, the present invention changes the voiceencoding rate and corresponding level of channel coding in order tomaximize the voice quality. For example, in environments where the C/Nratio is high, the system uses the highest possible vocoder rate andlowest possible amount of channel coding. In this situation, because ofthe low levels of noise and interference on the communication channel,there is little need for heavy channel coding to ensure thecommunication channel is sustained. However, as the C/N ratio begins todecrease, at the precise instant when curve 306 crosses curve 312, shownas intersection 310, the communication system of the present inventionchanges the vocoder rate and corresponding channel coding to the rateassociated with curve 312. In this manner, the highest possible level ofvoice quality is maintained, even though there is a higher level ofchannel coding present.

Similarly, as the voice quality of the system represented by curve 312drops to the level of the system represented by curve 318, shown atintersection 316, the communication system of the present inventionagain changes the vocoder rate and corresponding channel coding to therate associated with curve 318. In this manner, the voice quality forthe communication channel is always maximized.

In the event the channel noise and interference exceeds the maximumallowable level and results in the voice quality being sufficiently poorso as to pass below the threshold 324, shown at intersection 322, thecommunication channel is terminated. This terminated communicationchannel is perceived by the user as a “dropped call,” Once terminated,system 100 must be re-initialized and a communication channel must bere-established between the BS 102 and the MS 110.

Graph 300 has been divided into three (3) regions 308, 314, and 320,representing the maximized voice quality. A communication channel usingthe present invention will operate within each of these regions asneeded to maximize the voice quality. For example, for a communicationchannel which is initiated at a vocoder and channel coding rate inregion 314 corresponding to curve 312, a momentary decrease in the C/Nmay cause the system to switch to a vocoder and channel coding rate inregion 320 corresponding to curve 318. However, once the C/N returns toits original value, the system will shift back to the vocoder andchannel coding rate of region 314 corresponding to curve 312. In thismanner, system 100 may constantly move between regions 308, 314 and 320to maximize the voice quality of the communication channel.

Graph 300 has been shown to include three (3) separate curvesrepresenting three (3) different vocoder rates and corresponding levelsof channel coding. However, it should be appreciated that the selectionof three (3) vocoder rates is merely exemplary, and virtually any numberof vocoder rates may be used in the present invention. Moreover, in asystem of the present invention incorporating a vocoder having avariable vocoder rate, virtually any combination of vocoder rate andchannel coding may be accomplished within the system limits, rangingfrom a maximum vocoder rate with no channel coding, to minimum vocoderrate with maximum channel coding.

Referring now to FIG. 4, a diagrammatic representation of theconstruction of a GSM communication channel is shown and generallydesignated 400. Representation 400 includes a series of three (3) speechblocks 402, 404, and 406. Speech block 402 includes a channel codingportion 408 and a voice data portion 410. A speech block represents thedigital information which has been generated by the vocoder 206, 208 or210 and channel coder 216 of circuit 200. Accordingly, the digitalinformation within a speech block includes both the voice data andchannel coding which has been determined necessary for the reliabletransmission of the information. While FIG. 4 identifies a channelcoding portion 408 and a voice data portion 410 as separate portions ofspeech block 402, it is to be appreciated that such identification ismerely for discussion purposes, and the voice data is actuallyinterleaved with the channel coding to create a data stream having 228bits.

Speech blocks 402, 404 and 406 are each shown having different ratios ofchannel coding portions and voice data portions. More specifically,speech block 402 is shown having a larger proportion of channel coding408 to a smaller proportion 412 of voice coding 410. Speech block 404,on the other hand, has approximately an even proportion 418 of channelcoding 414 to voice coding 416. Speech block 406 has a larger proportion424 of channel coding 420 to voice coding 422. In any case, fromcomparing speech blocks 402, 404, and 406, it can be seen that theratios of channel coding to voice coding may change, and even thoughthree separate ratios have been shown in FIG. 4, virtually anyproportion 412, 418, and 424 may be implement with the presentinvention.

In addition to having a variable quantity of voice data and channelcoding, a speech block may also be encoded with a number of rate bits413. These rate bits 413 represent the particular vocoder rate withwhich the voice data is encoded. For example, in a communication systemwhere vocoder rates may be varied, rate bits 413 provide the necessaryvocoder rate information to successfully decode the voice data. In apreferred embodiment, the rate bits are positioned within the speechblock 402, but are not convolutionally encoded with the voice data andchannel coding. Rather, the rate bits 413 are “soft-coded” into thespeech block 402 such that they can be extracted without the need forconvolutionally decoding the speech block. The term “extract” in thepresent context may include convolutionally decoding, soft-decoding,hard-decoding, or any other manner of retrieving the digital informationfrom the data stream known in the art.

The “soft-coding” of the rate bits may be accomplished by placing aseries of bits within a particular location of the speech block. Forexample, rate bits 413 may be placed at bit positions 70, 71, and 72 ofspeech blocks 402, 404 and 406. By positioning the rate bits atconsistent locations within each of the speech blocks, it is notnecessary to decode the block to determine the value of the rate bits.Instead, the value of the bits in bit positions 70, 71 and 72 could bedetermined simply by scanning those bits in the serial bit stream.Additionally, it is possible to place the rate bits in more than onelocation within each speech block, providing for a measure of errorcorrection. For example, rate bits 413 could occur in three separatelocations within speech block 402, allowing the averaging of the bitswithin the three separate locations in order to provide the bestapproximation of the rate bits despite any transmissions errors.

In a preferred embodiment, rate bits 413 may represent a three-bitbinary value corresponding to eight distinct vocoder rates. Table 1below identifies such a table of eight distinct vocoder rates based uponthree rate bits. As can be seen from Table 1, the rate bits 413 may beassigned any vocoder rate within the vocoder range of the communicationsystem.

TABLE 1 Rate Bits for Corresponding Vocoder Rates Rate Bits Vocoder Rate(Level) Vocoder Rate (Kbits/s) 000 Level 1 3.0 Kbits/s 001 Level 2 4.0Kbits/s 010 Level 3 5.0 Kbits/s 011 Level 4 6.0 Kbits/s 100 Level 5 7.0Kbits/s 101 Level 6 8.0 Kbits/s 110 Level 7 9.0 Kbits/s 111 Level 8 10.0Kbits/s

There are three (3) rate bits identified in Table 1, however, the numberof rate bits may vary depending upon the total number of vocoder ratesavailable. For instance, if only two rates are available, a single bitwould be needed, with a bit value of “0” indicating one rate, and thebit value of “1” indicating the other rate. Similarly, if only fourrates were available, two rate bits would be needed, with the bit valuesof “00” indicating a first vocoder rate, bit values of “01” indicating asecond vocoder rate, bit values of “10” indicating a third vocoder rate,and bit values of “11” indicating a fourth vocoder rate.

Although Table 1 includes a series of eight (8) vocoder rates spaced 1Kbits/s apart, it is to be appreciated that it is not necessary for thevocoder rates to be evenly distributed. In fact, it would beadvantageous for the communication system of the present invention tohave a number of vocoder rates within the operating range mostfrequently experienced by the system. For example, if the communicationsystem noise and interference characteristics indicate that the vocoderrate would typically be 6 Kbits/s, then it might be advantageous toprovide several vocoder rates within the 5 to 7 Kbits/s region in orderto maximize the voice quality. In such an environment, a series of eight(8) vocoder rates might include the following vocoder rates: 4.0Kbits/s, 5.0 Kbits/s, 5.5 Kbits/s, 6.0 Kbits/s, 6.5 Kbits/s, 7.0Kbits/s, 8.0 Kbits/s, and 9.0 Kbits/s. Using these vocoder rates wouldallow the communication system to adjust the vocoder rate just slightlyin order to provide the finest possible voice quality during periods ofslight fluctuations in the channel noise and interference levels, whileretaining the ability to significantly change the vocoder rate forperiods of heavy channel noise and interference.

Once the voice data has been encoded with the necessary channel coding,and any rate coding, to form speech blocks 402, 404, and 406, eachspeech block is divided into four (4) sub-blocks. For example, “A”speech block 402 is split into sub-blocks “A₁” 432, “A₂” 434, “A₃” 436,and “A₄” 438. Likewise, “B” speech block 404 is split into sub-blocks“B₁” 440, “B₂” 442, “B₃” 444, and “B₄” 446, and “C” speech block 406 issplit into sub-blocks “C₁” 448, “C₂” 450, “C₃” 452, and “C₄” 454. Inthis manner, the 228 bit data stream in the speech block is broken intofour (4) sub-blocks of 57 bits each.

Using a combination of sub-blocks, a multi-frame 476 is constructedwhich includes a continual string of data frames, with each data framehaving eight (8) time slots 478, 480 and 482. As shown by mapping lines462 and 464 in FIG. 4, frame 478 is constructed from the “A₃” sub-block436 and the “B₁” sub-block 440. This combination of sub-blocks intoframes 478, 480 and 482 is called “frame-interleaving” and is intendedto create a more robust communication channel.

In addition to this frame-interleaving, the even bits within frame 478are comprised of the data bits of the “B₁” sub-block 440, and the oddbits within frame 478 are comprised of data bits of the “A₃” sub-block436. This even bit/odd bit combination is called “bit-interleaving” andresults in the distribution of a single speech block over fourcontiguous frames. This distribution provides for an improved faulttolerance for the communication system, and in circumstances where thenoise level and interference level are high, results in a more resilientcommunication channel.

In addition to the combination of sub-blocks 438 and 442, frame 480 isalso encoded with communication system specific coding. For example,using the GSM-based communication system of FIG. 1, frame 480 is encodedwith three (3) leading “tail bits” 482, a first “encoded voice” bitstream 484 of fifty-seven (57) bits, a single “flag” bit 486, atwenty-eight bit “training sequence” 488, a second “flag” bit 490, asecond “encoded voice” bit stream 492 of fifty-seven (57) bits, three(3) trailing “tail bits” 494, and an eight and one-quarter (8¼) bit“guard” period 496. The first and second “encoded voice” bit streams 484and 492 represent the encoded voice which was present in the “B₁”sub-block 440 and the “A₃” sub-block 436, which included both the voicedata, channel coding, and rate bits.

Because Doppler shift and multi-path echoes in system 100 can affect thereceived signal quality, each TDMA frame must include training sequence488, also called training bits. The receiver in system 100 comparesthese training bits with a known training pattern, and from this deducesthe transfer function of the propagation path. An adaptive filter isthen created within processor 214 to perform the inverse transferfunction, thus canceling any unacceptable distortion. This adaptivefiltering is well known in the art, and is thus not discussed in moredetail here.

Because of the frame-interleaving and bit-interleaving employed in thisGSM-based communication system 100, it is not possible to decode thevoice information without re-assembling the sub-blocks 432-454 fromsuccessive frames 478-482 in multi-frame 476. Consequently, it isnecessary for the digitally encoded voice information to be temporarilystored, such as by temporarily placing the encoded voice informationinto storage 215 of circuit 200. Once a sufficient number of frames hasbeen stored in memory storage 215, the sub-blocks are then re-assembledand the voice data is decoded from the re-constructed speech blocks,removing all channel coding, and sent through switch 248 to vocoders250, 252, and 254.

A full-rate GSM-based system would assign each time slot within a frameto a different user. For example, each of the eight (8) time slotswithin a frame would be assigned to eight (8) different users. In ahalf-rate GSM-based system, the frame and slot timing remains the same,but instead of a user being assigned a time slot in every frame, theuser is assigned a time slot in every other frame.

Operation

Communication Channel Metrics

The operation for the present invention includes the modification of thevocoder rate and level of channel coding to provide the best possiblevoice quality, while ensuring a reliable communication channel. In orderto determine the appropriate level of channel coding necessary toprovide reliable communication, a number of channel quality metrics areconsidered by the present invention. Defined generally, these channelquality metrics include characteristics of the communication channelwhich may be measured, and by continually measuring these channelquality metrics, an accurate evaluation of the channel quality may bemade.

One channel metric used to evaluate the quality of the communicationchannel is the uncoded Bit Error Rate (BER). The uncoded BER of acommunication channel is defined as the ratio of the number of bits in adata stream which are improperly demodulated to the total number of bitstransmitted. In general, a bit error is caused when the noise powerlevel in a communication system becomes comparable to the energy levelin each bit transmitted Consequently, in a system with a smallchannel-to-noise ratio (C/N), bit errors are more likely. Conversely, ina system with a large channel-to-noise ratio, bit errors are lesslikely. Thus, on a fundamental level, the rate of occurrence of biterrors, or the BER, provides an overall system quality metric.

An additional metric which may be used to evaluate the quality of acommunication channel is the RX Quality (RXQ) indicator. The RXQindicator as generally known in the industry is assigned a value by thenetwork, indicating the quality of the received signal based upon thecurrent BER. Table 2 below includes values for a typicalnetwork-determined BER with corresponding RXQ values. This table,however, represents an average received quality, and not aninstantaneous RXQ value.

TABLE 2 GSM Standards for RX Quality Metric RX Qual Corresponding BitError Rate Range of Actual BER (%) 0 Below 0.2 Below 0.1 1 0.2 to 0.40.26 to 0.30 2 0.4 to 0.8 0.51 to 0.64 3 0.8 to 1.6 1.0 to 1.3 4 1.6 to3.2 1.9 to 2.7 5 3.2 to 6.4 3.8 to 5.4 6  6.4 to 12.8  7.6 to 11.0 7above 12.8 above 15

The GSM standards for the RXQ of Table 1 is an average value measuredduring a predefined period of time. However, because the presentinvention contemplates an immediate response to a decrease in the RXQvalue, it is necessary to determine the RXQ metric on a block-by-blockbasis. This block-by-block calculation of RXQ′, for example, would bemade within the MS for the downlink, and within the BS for the uplink.

In the present invention, an RXQ′ metric is defined and is dynamicallymeasured by re-encoding the decoded voice data coming out of theconvolutional decoder, and comparing them against the received bits. TheRXQ′ value represents the number of bits different between the receivedbits and the re-encoded bits per block. The RXQ′ consequently provides acombined indication of bit error rate and receiver quality for eachblock.

Referring briefly to FIGS. 2 and 4, the determination of the RXQ′ metricis accomplished by decoding the voice data from a speech block 402within a received frame 480, and re-coding the voice data for comparisonto the encoded received data. The determination of the RXQ′ metric takesplace within circuit 200 by receiving a transmitted frame 480 andpassing the frame through transmit/receive switch 224 to intermediatefrequency (IF) amplifier and mixer 240, through ADC 242 and equalizer244, to channel decoder 246. In channel decoder 246, the frame 480 isdecoded to the original speech block which is then passed to storage 215for later use. Following storage of the original speech block, allchannel coding is removed to recover the original voice data which mayalso be stored in storage 215, or passed on through switch 248 tovocoders 250, 252 or 254 for conversion to audio.

Once the original voice data is recovered from channel decoder 246, thenow-decoded voice data is then re-encoded through a convolutional codingprocess identical to that of the channel encoder 216 to exactlyre-create the original coded speech block. This re-encoding may beaccomplished using channel decoder 246, or the voice data may be passedthrough a separate channel coder 247. By comparing the original speechblock stored in storage 215 with the newly re-coded speech block fromchannel coder 247, an estimated bit-error-rate may be determined. Forexample, by comparing the received speech block with the re-coded speechblock, the existence of any error-correction which has taken placewithin channel decoder will become apparent. Consequently, this dynamicmethod of error detection is considerably more sensitive than otherestimates of the BER, and may be done on a block-by-block basis.

An additional metric, SRXQ, is defined as the weighed sum of prior RXQ′measurements. The SRXQ metric is intended to introduce some history intothe vocoder rate decision making process based on the receiver quality.In one embodiment, the RXQ′ measurements for the prior five (5) blocksare considered in the SRXQ measurement. The prior RXQ′ measurements areweighted in accordance with the following equation:SRXQ=SUM(2^(K−1))(RXQ′(K+4));

where K=−4, −3, −2, −1, and 0, and where RXQ′(0) is the measured valuefor the most recent block.

An alternative channel quality metric, Frame Erase (FE), may be used todetermine the overall quality of the channel. The FE metric representsthe number of frames which have been determined to be corrupted, andconsequently not used in re-generating the original voice data. In otherwords, the FE metric represents a count of the number of frames whichhave been erased per unit time. The decision to erase a frame may bemade using a number of criterion. In a present embodiment, thedetermination to erase a frame is made based on thecyclic-redundancy-checking (CRC), also generally known as a “parity”check. Based on a CRC value which is decoded from the received frame, aframe is either used or discarded, avoiding the use of a frame which mayhave been improperly decoded or otherwise corrupted.

System Operation

Referring now to FIG. 5, a state diagram is shown and generallydesignated 500. State diagram 500 represents the changes in vocoder andchannel coding rates in response to changes in the communication systemenvironment. For discussion purposes, it is assumed that thecommunication system is initially experiencing a high carrier-to-noiseratio (C/N), and thus the system is initially in state 502 having arelatively high vocoder rate of 8 Kbits/s, with a correspondingly lowlevel of channel coding. In other words, state 502 is used in low-noiseenvironments, such as where the carrier-to-interference ration (C/I)exceeds 19 dB, wherein the majority of digital information with a speechblock may be voice data. System 100 will remain in state 502 so long asthe FE metric remains at zero (0), as indicated by control path 508.This results in a communication system having a superior voice quality.

In the event that a frame is erased resulting in the FE metric becomingnon-zero, the BER is computed to determine whether it meets or exceeds athreshold value. In the present embodiment, this threshold value is onepercent (1%), meaning that if more than one bit out of a total bitstream of one hundred (100) bits is erroneous, the threshold is met orexceeded. Once the FE metric becomes non-zero and the BER is above theone percent (1%) threshold, the system changes to state 504 via controlpath 510.

State 504 is used in environments exhibiting moderate levels of noiseand interference, and combines a mid-range vocoder rate of 6 Kbits/swith a moderate level of channel coding. In the current example, thevocoder and channel coding rate will remain at the mid-range of state504 so long as the BER is greater-than-or-equal-to one percent (1%), andless than five percent (5%) (1% £ BER <5%). In this state, typicallywhere the C/I is between 10 and 19 dB, the communication channelexhibits a reasonably good voice quality.

If after a period of time the communication environment improves and theFE metric returns to zero (0) and the BER becomes less than one percent(1%), the system returns to state 502 via control path 512. On the otherhand, in the event the system environment becomes more noisy and thechannel-to-noise ratio (C/N) becomes smaller, the FE metric will likelyincrease. If the FE metric increases to equal or exceed 5, and the BERmetric is greater-than-or-equal-to 5 percent (5%), (FE>5 and 5% £ BER)the system passes to state 506 via control path 516. In this state, ahigher degree of channel coding is implemented resulting in acorresponding lower vocoder rate of 4 Kbits/s. According to control path520, the system will remain in state 506 so long as the BER isgreater-than-or-equal-to 5% (5% £ BER), typically when the C/I isbetween 4 to 10 dB.

When the system is in state 506, and the communication environmentimproves causing the FE metric to decrease to zero (0) and the BERmetric to decrease to less than five percent (5%), than thecommunication system will change to state 504 according to control path518, thereby decreasing the level of channel coding and improving thevoice quality of the system.

In the event the communication system is in state 506 and the FE and BERmetrics continue to increase, the communication system may eventuallydiscontinue the communication channel resulting in a “dropped call.” Inthe present embodiment, the communication channel will be discontinuedwhen the FE and BER rates exceed 20 and ten percent (10%), respectively,for example.

In order to ensure the proper operation of the system of the presentinvention, it is necessary that the metrics evaluated for determinationof the system control between various states 502, 504, and 506 include ameasure of hysteresis. For example, if no hysteresis were to be includedbetween states 502 and 504, it would be possible for the system tooscillate rapidly between the two states, resulting in a vocoder rateand level of channel coding which varies from frame to frame. Althoughthis continual vocoder change is possible with the system of the presentinvention, it is unnecessary and may result in an inefficient use ofsystem resources.

The discussion of the various FE and BER values set forth above isintended as one example of a preferred embodiment having three (3)different vocoder and channel coding rates. The FE and BER values setforth are merely exemplary, and any number of alternative FE and BERvalues may be chosen and implemented. The threshold values for the FEand BER values may be treated as system parameters, and may change fordifferent vocoders. Also, FIG. 5 shows a state diagram with three (3)states, however, any number of states may be created within the presentinvention.

Mobile Station Control of Downlink Rate

Referring now to FIG. 6, a flow chart representing the operation of thecommunication system of the present invention is shown and generallydesignated 600. In general, this configuration includes the MSdetermining the proper downlink vocoder rate and level of channelcoding. Following this determination, the MS then transmits thenecessary rate information to the BS.

Flow chart 600 begins with first step 602 which includes reception of aradio-frequency frame at the MS. Following receipt of the frame at theMS, the soft-coded rate bits are extracted from the frame data in step604. In a preferred embodiment of the present invention and as discussedabove in conjunction with FIG. 4, these soft-coded rate bits may includethree (3) bits of rate information that can identify up to eight (8)different vocoder and channel coding rates. The frame data is thenconvolutionally decoded to yield the original speech block in step 605.

Using the appropriate vocoder and channel coding rate informationextracted in step 604, the speech block is then decoded to recreate theoriginal voice data in step 606. In this manner, the MS may receive aframe containing voice data encoded with virtually any vocoder rate, andthe frame may be successfully decoded to the original voice data becauseall relevant vocoder rate information is transmitted within the frame inthe form of soft-coded bits.

In order to provide the best possible voice communication channel, theMS determines the channel quality metrics discussed above, such as FE,BER and RXQ, in step 608. The MS also calculates the SRXQ value in step610 and, based upon the results of the measured and calculated metrics,determines the vocoder rate for optimal voice quality in step 612. In apreferred embodiment of the present invention, the rate bitscorresponding to the new vocoder and channel coding rate are determinedfrom a look-up table. Once the vocoder and channel coding rate isdetermined, the MS transmits a frame with the new downlink vocoder rateconvolutionally coded into the frame in step 614. Uplink 626 representsthe transmission of a frame from the MS to the BS.

In step 616, the BS receives the frame containing theconvolutionally-coded downlink vocoder rate for the next downlinktransmission. Because it is not necessary to know the downlink vocoderrate in order to decode the uplink transmission, the downlink vocoderrate may be convolutionally encoded instead of soft-coded.

In step 618, the BS decodes the received frame from the MS yielding thenew downlink vocoder rate bits. These vocoder rate bits are used todetermine, using a look-up table or the like, the new downlink vocoderrate. Using that newly determined vocoder rate, the BS encodes the voicedata in step 619 for transmission to the MS. In step 620, the BStransmits the frame containing the convolutionally encoded voice dataand coded downlink vocoder rate bits to the MS. Downlink 628 representsthe transmission of a frame from the BS to the MS.

Importantly, each downlink message includes as soft-coded bits the rateinformation related to the speech block. This is so because there existsa possibility that a frame may become corrupted and no longer readable.This corruption may create a situation wherein the MS may havetransmitted a message frame in the uplink changing the downlink vocoderrate, and that frame was not successfully received by the BS. If thisoccurs, the MS would expect to receive a frame having a new vocoderrate, while the frame actually received would be encoded at the oldrate. Additionally, in circumstances involving discontinuoustransmissions (DTX), such as when the MS is not transmitting to savebattery power, the channel characteristics and corresponding vocoderrate information could change significantly between transmitted frames.Consequently, in order to avoid such miscommunication, each speech blockis soft-coded with the rate information necessary to decode the speechblock.

In a preferred embodiment of the present invention as shown in FIG. 6,steps within sequence 600 identified by bracket 622 are performed withinthe MS, and steps within sequence 600 identified by bracket 624 areperformed within the BS.

In any one cycle of uplink-downlink transmissions shown in FIG. 6, boththe BS and the MS will inform the other of the appropriate vocoder ratesfor the transmitted message. For example, in an uplink frame containingconvolutionally-coded rate bits for the next downlink frame, soft-codedrate bits will be present which will tell the BS what vocoder rate touse in decoding the uplink frame. Similarly, in a downlink framecontaining convolutionally-coded rate bits for the next uplink frame,soft-coded rate bits will be present which will tell the MS what vocoderrate to use in decoding that downlink frame.

In the communication system of the present invention, it has been termedthat vocoder rate bits which are not convolutionally encoded are“soft-coded” into the speech block, and the vocoder rate bits which areconvolutionally encoder are “hard-coded” into the speech block. As analternative terminology, the vocoder rate information which isconvolutionally encoded into a speech block could also be considered an“inside” rate, as the vocoder rate information is within theconvolutional coding. Vocoder rate information which is soft-coded intothe speech-block is considered an “outside” rate, as the vocoder rateinformation is outside the convolutional coding.

Base Station Control of Downlink

Referring now to FIG. 7, a flow chart representing the operation of analternative embodiment of the communication system of the presentinvention is shown and generally designated 700. In general, thisconfiguration includes the MS monitoring a series of channel metrics andrelaying this metric information to the BS for determining the properdownlink vocoder rate and level of channel coding. Following thisdetermination, the BS then transmits the soft-coded rate bits to the MSwith the following frame.

In first step 702, the MS receives a frame with soft-coded rate bits. Instep 703, the MS extracts the soft-coded rate bits from the frame, andusing a look-up table or the like, determines the appropriate downlinkvocoder rate and level of channel coding. In step 704, using this rateinformation, the MS decodes the frame, yielding a speech block. In step706, the vocoders are set to the appropriate rate and this speech blockis decoded to re-create the original voice in the speech block.

During the decoding process, the MS is determining the channel qualityof the communication system. For example, quality metrics such as FE andRXQ may be determined in step 708. Following the determination of FE andRXQ, a quantized vocoder value is determined in step 710 which reflectsthe current communication channel quality. Referring ahead briefly toFIG. 8, a quantization table is shown and generally designated 800.Quantization table 800 includes both the FE metric 802 and the RXQmetric 804 which are measured at the MS, and lists a number ofnon-uniform quantization values for each. These RXQ′ values are themid-range of the transmitted quantization levels, and represent a rangeof RXQ′ metric values. Since the FE and RXQ′ are both associated withthe receiver performance, the quantization of RXQ′ is based on the valueof FE to effectively quantize RXQ′ into eight levels. By locating thecurrent measured values of both the FE and RXQ on the quantizationtable, a series of three (3) quantization bits are identified. Forinstance, for a FE value of 1 and a RXQ value of 25, quantization bits1-0-0 are selected. Once the quantization bits are selected, in step 712a frame is transmitted from the MS to the BS with the quantization bitsfully encoded in the speech block. Uplink 730 represents thetransmission of a frame from the MS to the BS.

In step 714, the frame is received at the BS with the quantization bitsfully encoded. This frame is decoded in step 716 to yield the originalquantization bits. Referring briefly to FIG. 9, a quantization table 900is shown which provides a look-up table to reconstruct the FE and RXQ′values from the received quantization bits. For example, forquantization bits 1-0-0, a FE metric value 904 of “1” and an RXQ metricvalue 906 of “22.” These metrics derived from the quantization bits arethen used to calculate the SRXQ metric in step 718. Based upon thequantization bits and the results of the SRXQ calculation, a new vocoderrate is determined in step 720 by the BS. In step 722, voice data forthe next speech block is encoded using the new vocoder rate, with thenew vocoder rate bits being soft-coded into the speech block resultingin a new frame. This new frame is then transmitted from the BS to the MSin step 724. Downlink 732 represents the transmission of a frame fromthe BS to the MS.

Base Station Control of Uplink

In addition to the rate bits which are exchanged between the MS and theBS to govern the downlink vocoder and channel coding rate, the rate bitscorresponding to the operation of the uplink are also exchanged. This isaccomplished by the BS analyzing similar channel quality metrics whichare used to determine the appropriate downlink vocoder rate as discussedin conjunction with FIG. 6.

Referring now to FIG. 10, a flow chart representing the operation of analternative embodiment of the communication system of the presentinvention is shown and generally designated 1000. In general, thisconfiguration includes the BS monitoring a series of channel metricsdetermines the proper uplink vocoder rate and level of channel coding.Following this determination, the MS then transmits the soft-coded ratebits to the BS with the following frame.

Flow chart 1000 begins with first step 1002 which includes reception ofa radio-frequency frame at the BS. Following receipt of the frame at theBS, the soft-coded rate bits are extracted from the frame data in step1004. In a preferred embodiment of the present invention and asdiscussed above in conjunction with FIG. 4, these soft-coded rate bitsmay include three (3) bits of rate information that can identify up toeight (8) different vocoder and channel coding rates. The frame data isthen convolutionally decoded to yield the original speech block in step1006.

Using the appropriate vocoder and channel coding rate informationextracted in step 1004, the speech block is then decoded to recreate theoriginal voice data in step 1008. In this manner, the BS may receive aframe containing voice data encoded with virtually any vocoder rate, andthe frame may be successfully decoded to the original voice data becauseall relevant vocoder rate information is transmitted within the frame inthe form of soft-coded bits.

In order to provide the best possible voice communication channel, theBS determines the channel quality metrics discussed above, such as FE,BER and RXQ, in step 1010. The BS also calculates the SRXQ value in step1012 and, based upon the results of the measured and calculated metrics,determines the vocoder rate for optimal voice quality in step 1014. In apreferred embodiment of the present invention, the rate bitscorresponding to the new vocoder and channel coding rate are determinedfrom a look-up table. Once the vocoder and channel coding rate isdetermined, the BS transmits a frame with the new uplink vocoder rateconvolutionally coded into the frame in step 1016. Downlink 1018represents the transmission of a frame from the BS to the MS.

In step 1020, the MS receives the frame containing theconvolutionally-coded uplink vocoder rate for the next downlinktransmission. Because it is not necessary to know the uplink vocoderrate in order to decode the uplink transmission, the uplink vocoder ratemay be convolutionally encoded instead of soft-coded.

In step 1022, the MS decodes the received frame from the BS yielding thenew uplink vocoder rate bits. These vocoder rate bits are used todetermine, using a look-up table or the like, the new uplink vocoderrate. Using that newly determined vocoder rate, the MS encodes the voicedata in step 1024 for transmission to the BS. In step 1026, the MStransmits the frame containing the convolutionally encoded voice dataand soft-coded uplink vocoder rate bits to the BS. Uplink 1028represents the transmission of a frame from the MS to the BS.

Importantly, each uplink message includes as soft-coded bits the rateinformation related to the speech block. This soft-coding enables the BSto properly decode the speech block without knowing in advance thevocoder rate. This is particularly advantageous because there exists apossibility that a frame may become corrupted and no longer readable.This corruption may create a situation wherein the BS may havetransmitted a message frame in the downlink changing the uplink vocoderrate, and that frame was not successfully received by the MS. If thisoccurs, the BS would expect to receive a frame having a new vocoderrate, while the frame actually received would be encoded at the oldrate. Additionally, in circumstances involving discontinuoustransmissions (DTX), such as when the BS is not continuouslytransmitting, the channel characteristics and corresponding vocoder rateinformation could change significantly between transmitted frames.Consequently, in order to avoid such mis-communication, each speechblock is soft-coded with the rate information necessary to decode thespeech block.

In a preferred embodiment of the present invention as shown in FIG. 10,steps within sequence 1000 identified by bracket 1030 are performedwithin the BS, and steps within sequence 1000 identified by bracket 1032are performed within the MS.

System Performance

The communication system of the present invention provides for the blockand bit interleaving thereby minimizing the disruption to thecommunication link caused by channel noise, interference, and droppedframes. In addition to such redundancy, the vocoder rate informationwhich is either hard-coded within the frame or soft-coded outside theframe, may also be repetitive. Such repetition will further enhance theresilience of the communication system of the present invention.Redundancy of the vocoder rate information, or rate bits, may beaccomplished by repeating the bits in several locations within thespeech frame, as mentioned above in conjunction with FIG. 4.

Like traditional GSM-based communication systems, the communicationsystem of the present invention provides for the transfer, or“hand-off,” of a MS from one BS to another BS in a different cell. Insuch a hand-off, it would not be necessary to provide the new BS withany special rate information via the communication link 108 as allnecessary vocoder rate information is presented in each frametransmitted from the MS.

The present invention may be implemented in either a full-rate orhalf-rate GSM-based communication system. The encoding and transmissionof the vocoder rate information between the BS and MS in both the fulland half-rate system would be identical.

In addition to the modification of the vocoder rate and channel codingas discussed above, the power level of the transmissions may also bemodified in order to provide the best possible voice quality. Forexample, in FIGS. 8 and 9, rate bits 806 and 902 may take intoconsideration, in addition to the FE and RXQ′ metrics, a metric relatedto the power level of the transmission. In such a situation, the BS mayadjust the vocoder rate and channel coding, while at the same timeadjusting the BS transmit power to minimize the BER or FE, resulting inbetter voice quality.

While the present invention has been discussed at length with respect tothe transmission of voice data between a BS and a MS, it should beappreciated that any digital data may be communicated in a similarmanner. In fact, because other types of digital data may not bedependent upon the audio sampling rates, a much higher data rate may beachieved using the present invention, and is fully contemplated herein.

Other Embodiments

While there have been shown what are presently considered to bepreferred embodiments of the invention, it will be apparent to thoseskilled in the art that various changes and modifications can be madeherein without departing from the scope and spirit of the invention asdefined by the appended claims and their equivalents.

1. A method of wireless communication for use in a wirelesscommunication system, the method comprising: receiving first-directionlink voice information, first-direction link vocoder rate bits andinformation for a vocoder rate of second-direction link voiceinformation through a first-direction link channel, wherein thefirst-direction link vocoder rate bits indicate a vocoder rate used by aremote station to encode the first-direction link voice information;decoding the first-direction link voice information using thefirst-direction link vocoder rate bits; measuring at least one channelquality metric of the first-direction link channel; generatinginformation bits for a new vocoder rate of the first-direction linkvoice information corresponding to the at least one first-direction linkchannel quality metric; encoding the second-direction voice informationusing the received information for the vocoder rate of thesecond-direction link voice information; and transmitting, through asecond-direction link channel, the encoded second-direction link voiceinformation, second-direction link vocoder rate bits and the generatedinformation bits, wherein the second-direction link vocoder rate bitsare for a presently applied vocoder rate in the encoding.
 2. The methodof claim 1, further comprising soft-coding the first-direction linkvocoder rate bits and the second-direction link vocoder rate bits. 3.The method of claim 1, wherein the first-direction link channel is adownlink channel and the second-direction link channel is an uplinkchannel.
 4. The method of claim 3, wherein the remote station is a basestation.
 5. The method of claim 1, wherein the first-direction linkchannel is an uplink channel and the second-direction link channel is adownlink channel.
 6. The method of claim 5, wherein the remote stationis a mobile station.
 7. The method of claim 1, wherein a higher vocoderrate is used when a carrier to noise ratio (C/N) increases and a lowervocoder rate is used when C/N decreases.
 8. An apparatus for use in awireless communication system, the apparatus comprising: a receiverconfigured to receive first-direction link voice information,first-direction link vocoder rate bits and information for a vocoderrate of second-direction link voice information through afirst-direction link channel, the first-direction link vocoder rate bitsused by a remote station to encode the first-direction link voiceinformation; a decoder configured to decode the first-direction linkvoice information using the first-direction link vocoder rate bits; aprocessor configured to measure at least one channel quality metric ofthe first-direction link channel, generate information bits for a newvocoder rate of the first-direction link voice information correspondingto the at least one first-direction link channel quality metric, andencode the second-direction voice information using the receivedinformation for the vocoder rate, of the second-direction link voiceinformation; and a transmitter configured to transmit the encodedsecond-direction link voice information, second-direction link vocoderrate bits and the generated information bits through a second-directionlink channel, wherein the second-direction link vocoder rate bits arefor a presently applied vocoder rate in the encoding.
 9. The apparatusof claim 8, wherein the first-direction link vocoder rate bits and thesecond-direction link vocoder rate bits are soft-coded.
 10. Theapparatus of claim 8, wherein the first-direction link channel is adownlink channel and the second-direction link channel is an uplinkchannel.
 11. The apparatus of claim 10, wherein the remote station is abase station.
 12. The apparatus of claim 8, wherein the first-directionlink channel is an uplink channel and the second-direction link channelis a downlink channel.
 13. The apparatus of claim 12, wherein the remotestation is a mobile station.
 14. The apparatus of claim 8, wherein ahigher vocoder rate is used when a carrier to noise ratio (C/N)increases and a lower vocoder rate is used when C/N decreases.
 15. Themethod of claim 1, wherein the first-direction link voice information,the first-direction link vocoder rate bits and the information for thevocoder rate of the second-direction link voice information are receivedthrough a first-direction link data frame.
 16. The method of claim 1,wherein the encoded second-direction link voice information, thesecond-direction link vocoder rate bits and the generated informationbits are transmitted through a second-direction link data frame.
 17. Theapparatus of claim 8, wherein the first-direction link voiceinformation, the first-direction link vocoder rate bits and theinformation for the vocoder rate of the second-direction link voiceinformation are received through a first-direction link data frame. 18.The apparatus of claim 8, wherein the encoded second-direction linkvoice information, the second-direction link vocoder rate bits and thegenerated information bits are transmitted through a second-directionlink data frame.